7 Sep
2009
7 Sep
'09
11:09 a.m.
Hello,
Does kamailio support PRACK method ?
If yes, how?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
7 Sep
7 Sep
11:41 a.m.
On Montag, 7. September 2009, BERGANZ François wrote:
Does kamailio support PRACK method ?
Hi François,
yes. Inaki already explained it nicely here:
http://www.kamailio.org/pipermail/users/2008-June/017695.html
Regards,
Henning
11:47 a.m.
BERGANZ François wrote:
Does kamailio support PRACK method ?
If yes, how?
Well, of course it does. Kamailio supports any method, including any
methods you yourself invent.
PRACKs are routed like any other non-INVITE request within a dialog. In
other words, they should be covered by loose_route().
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
10 Aug
10 Aug
12:37 p.m.
New subject: Puzzling - INVITEs not making it to the fones
I'm seeing some *very* puzzling behavior from a bunch of our clients. Basically, the
INVITE never makes it to the fone, it looks like the upstream router/modem just swallows
it. The weird thing is, tweaking the username slightly resolves the issue, for a while at
least. And then, it stops again. Anyone seen anything like this before? Any ideas? Am
I missing something ridiculously obvious?
Fwiw, I've seen this behavior in setups with *no* SIP ALG, and *no* firewall. That
said, it really *does* look like a firewalling issue, doesnt it?
Here is a typical INVITE that is sent out from Kamailio. The username is mahesh.test204.
The "fix" is to simply change the name to "mahesh.test205", or
"mahesh.test20", or "ahesh.test204", etc.
Is there something obviously wrong with the INVITE perhaps? Bad format?
INVITE sip:mahesh.test204@173.101.106.53:5060;rinstance=e2f54e94d29118d7;transport=UDP
SIP/2.0.
Record-Route: <sip:74.217.82.147;lr;ftag=X8BN4N8vtXKej;nat=yes>.
Via: SIP/2.0/UDP 74.217.82.147;branch=z9hG4bK8e0c.0ded64d5.0.
Via: SIP/2.0/UDP
74.217.82.232;received=74.217.82.232;rport=5060;branch=z9hG4bK3DtFg45Qetv9F.
Max-Forwards: 69.
From: "7732206484" <sip:2065771035@74.217.82.232>;tag=X8BN4N8vtXKej.
To: <sip:mahesh.test204@a5prodsbc.aptela.com>.
Call-ID: 67108c50-1f0d-122e-17b1-001a647938bc.
CSeq: 382107 INVITE.
Contact: <sip:mod_sofia@74.217.82.232:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
PUBLISH, SUBSCRIBE.
Supported: precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 246.
Subject: Local.
Remote-Party-ID: "7732206484"
<sip:2065771035@74.217.82.147>;party=calling;id-type=subscriber;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1281422601 1281422602 IN IP4 74.217.82.232.
s=FreeSWITCH.
c=IN IP4 74.217.82.232.
t=0 0.
m=audio 13742 RTP/AVP 0 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.
Mahesh Paolini-Subramanya | CTO | mahesh(a)aptela.com | 703.386.1500 Ext. 9100
2250 Corporate Park Drive | Suite 150 | Herndon, VA | www.aptela.com
Check out our Blog | Follow us on Twitter | Refer a Friend
10:43 p.m.
New subject: Puzzling - INVITEs not making it to the fones
Hello,
On 8/10/10 12:37 PM, Mahesh Paolini-Subramanya wrote:
I'm seeing some *very* puzzling behavior from a
bunch of our clients.
Basically, the INVITE never makes it to the fone, it looks like the
upstream router/modem just swallows it. The weird thing is, tweaking
the username slightly resolves the issue, for a while at least. And
then, it stops again. Anyone seen anything like this before? Any
ideas? Am I missing something ridiculously obvious?
Fwiw, I've seen this behavior in setups with *no* SIP ALG, and *no*
firewall. That said, it really *does* look like a firewalling issue,
doesnt it?
if the request is sent by kamailio but not reaching the phone, then is
something in the middle inspecting SIP and discarding (big brother?). Is
it a particular type of phone, or any phone? At first sight, the request
seems just fine. I cannot think of something else than alg/firewall in
the middle, unless the phone itself discards it.
Cheers,
Daniel
Here is a typical INVITE that is sent out from Kamailio. The username
is /mahesh.test204/. The "fix" is to simply change the name to
"mahesh.test205", or "mahesh.test20", or "ahesh.test204",
etc.
Is there something obviously wrong with the INVITE perhaps? Bad format?
INVITE
sip:mahesh.test204@173.101.106.53:5060;rinstance=e2f54e94d29118d7;transport=UDP
SIP/2.0.
Record-Route: <sip:74.217.82.147;lr;ftag=X8BN4N8vtXKej;nat=yes>.
Via: SIP/2.0/UDP 74.217.82.147;branch=z9hG4bK8e0c.0ded64d5.0.
Via: SIP/2.0/UDP
74.217.82.232;received=74.217.82.232;rport=5060;branch=z9hG4bK3DtFg45Qetv9F.
Max-Forwards: 69.
From: "7732206484" <sip:2065771035@74.217.82.232>;tag=X8BN4N8vtXKej.
To: <sip:mahesh.test204@a5prodsbc.aptela.com>.
Call-ID: 67108c50-1f0d-122e-17b1-001a647938bc.
CSeq: 382107 INVITE.
Contact: <sip:mod_sofia@74.217.82.232:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: precondition, path, replaces.
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 246.
Subject: Local.
Remote-Party-ID: "7732206484"
<sip:2065771035@74.217.82.147>;party=calling;id-type=subscriber;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1281422601 1281422602 IN IP4 74.217.82.232.
s=FreeSWITCH.
c=IN IP4 74.217.82.232.
t=0 0.
m=audio 13742 RTP/AVP 0 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.
Mahesh Paolini-Subramanya | CTO | mahesh(a)aptela.com
<mailto:mahesh@aptela.com> | 703.386.1500 Ext. 9100
2250 Corporate Park Drive | Suite 150 | Herndon, VA | www.aptela.com
<http://www.aptela.com/>
Check out our Blog <http://www.aptela.com/blog> | Follow us on Twitter
<http://twitter.com/Aptela> | Refer a Friend
<http://www.aptela.com/referral>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com/
5001
days inactive
5338
days old
4 comments
5 participants
participants (5)
-
Alex Balashov -
BERGANZ François -
Daniel-Constantin Mierla -
Henning Westerholt -
Mahesh Paolini-Subramanya